/*
 * Copyright (C) 2014 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#pragma once

#include <math.h>
#include <stdint.h>

namespace cocos2d {

// AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original
// audio sample rate and the target rate when downsampling,
// as permitted in the audio framework, e.g. AudioTrack and AudioFlinger.
// In practice, it is not recommended to downsample more than 6:1
// for best audio quality, even though the audio framework permits a larger
// downsampling ratio.
// REFINE: replace with an API
#define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256

// AUDIO_RESAMPLER_UP_RATIO_MAX is the maximum suggested ratio between the
// original audio sample rate and the target rate when upsampling.  It is
// loosely enforced by the system. One issue with large upsampling ratios is the
// approximation by an int32_t of the phase increments, making the resulting
// sample rate inexact.
#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536

// AUDIO_TIMESTRETCH_SPEED_MIN and AUDIO_TIMESTRETCH_SPEED_MAX define the min
// and max time stretch speeds supported by the system. These are enforced by
// the system and values outside this range will result in a runtime error.
// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might
// be narrower than the ones specified here AUDIO_TIMESTRETCH_SPEED_MIN_DELTA is
// the minimum absolute speed difference that might trigger a parameter update
#define AUDIO_TIMESTRETCH_SPEED_MIN 0.01f
#define AUDIO_TIMESTRETCH_SPEED_MAX 20.0f
#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f
#define AUDIO_TIMESTRETCH_SPEED_MIN_DELTA 0.0001f

// AUDIO_TIMESTRETCH_PITCH_MIN and AUDIO_TIMESTRETCH_PITCH_MAX define the min
// and max time stretch pitch shifting supported by the system. These are not
// enforced by the system and values outside this range might result in a pitch
// different than the one requested. Depending on the
// AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
// the ones specified here.
// AUDIO_TIMESTRETCH_PITCH_MIN_DELTA is the minimum absolute pitch difference
// that might trigger a parameter update
#define AUDIO_TIMESTRETCH_PITCH_MIN 0.25f
#define AUDIO_TIMESTRETCH_PITCH_MAX 4.0f
#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
#define AUDIO_TIMESTRETCH_PITCH_MIN_DELTA 0.0001f

// Determines the current algorithm used for stretching
enum AudioTimestretchStretchMode : int32_t {
  AUDIO_TIMESTRETCH_STRETCH_DEFAULT = 0,
  AUDIO_TIMESTRETCH_STRETCH_SPEECH = 1,
  // REFINE: add more stretch modes/algorithms
};

// Limits for AUDIO_TIMESTRETCH_STRETCH_SPEECH mode
#define TIMESTRETCH_SONIC_SPEED_MIN 0.1f
#define TIMESTRETCH_SONIC_SPEED_MAX 6.0f

// Determines behavior of Timestretch if current algorithm can't perform
// with current parameters.
// FALLBACK_CUT_REPEAT: (internal only) for speed <1.0 will truncate frames
//    for speed > 1.0 will repeat frames
// FALLBACK_MUTE: will set all processed frames to zero
// FALLBACK_FAIL:  will stop program execution and log a fatal error
enum AudioTimestretchFallbackMode : int32_t {
  AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT = -1,
  AUDIO_TIMESTRETCH_FALLBACK_DEFAULT = 0,
  AUDIO_TIMESTRETCH_FALLBACK_MUTE = 1,
  AUDIO_TIMESTRETCH_FALLBACK_FAIL = 2,
};

struct AudioPlaybackRate {
  float mSpeed;
  float mPitch;
  enum AudioTimestretchStretchMode mStretchMode;
  enum AudioTimestretchFallbackMode mFallbackMode;
};

static const AudioPlaybackRate AUDIO_PLAYBACK_RATE_DEFAULT = {
    AUDIO_TIMESTRETCH_SPEED_NORMAL, AUDIO_TIMESTRETCH_PITCH_NORMAL,
    AUDIO_TIMESTRETCH_STRETCH_DEFAULT, AUDIO_TIMESTRETCH_FALLBACK_DEFAULT};

static inline bool isAudioPlaybackRateEqual(const AudioPlaybackRate &pr1,
                                            const AudioPlaybackRate &pr2) {
  return fabs(pr1.mSpeed - pr2.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
         fabs(pr1.mPitch - pr2.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA &&
         pr1.mStretchMode == pr2.mStretchMode &&
         pr1.mFallbackMode == pr2.mFallbackMode;
}

static inline bool
isAudioPlaybackRateValid(const AudioPlaybackRate &playbackRate) {
  if (playbackRate.mFallbackMode == AUDIO_TIMESTRETCH_FALLBACK_FAIL &&
      (playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_SPEECH ||
       playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_DEFAULT)) {
    // test sonic specific constraints
    return playbackRate.mSpeed >= TIMESTRETCH_SONIC_SPEED_MIN &&
           playbackRate.mSpeed <= TIMESTRETCH_SONIC_SPEED_MAX &&
           playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN &&
           playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX;
  } else {
    return playbackRate.mSpeed >= AUDIO_TIMESTRETCH_SPEED_MIN &&
           playbackRate.mSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX &&
           playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN &&
           playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX;
  }
}

// REFINE: Consider putting these inlines into a class scope

// Returns the source frames needed to resample to destination frames.  This is
// not a precise value and depends on the resampler (and possibly how it handles
// rounding internally). Nevertheless, this should be an upper bound on the
// requirements of the resampler. If srcSampleRate and dstSampleRate are equal,
// then it returns destination frames, which may not be true if the resampler is
// asynchronous.
static inline size_t sourceFramesNeeded(uint32_t srcSampleRate,
                                        size_t dstFramesRequired,
                                        uint32_t dstSampleRate) {
  // +1 for rounding - always do this even if matched ratio (resampler may use
  // phases not ratio) +1 for additional sample needed for interpolation
  return srcSampleRate == dstSampleRate
             ? dstFramesRequired
             : size_t((uint64_t)dstFramesRequired * srcSampleRate /
                          dstSampleRate +
                      1 + 1);
}

// An upper bound for the number of destination frames possible from srcFrames
// after sample rate conversion.  This may be used for buffer sizing.
static inline size_t destinationFramesPossible(size_t srcFrames,
                                               uint32_t srcSampleRate,
                                               uint32_t dstSampleRate) {
  if (srcSampleRate == dstSampleRate) {
    return srcFrames;
  }
  uint64_t dstFrames = (uint64_t)srcFrames * dstSampleRate / srcSampleRate;
  return dstFrames > 2 ? dstFrames - 2 : 0;
}

static inline size_t sourceFramesNeededWithTimestretch(uint32_t srcSampleRate,
                                                       size_t dstFramesRequired,
                                                       uint32_t dstSampleRate,
                                                       float speed) {
  // required is the number of input frames the resampler needs
  size_t required =
      sourceFramesNeeded(srcSampleRate, dstFramesRequired, dstSampleRate);
  // to deliver this, the time stretcher requires:
  return required * (double)speed + 1 +
         1; // accounting for rounding dependencies
}

// Identifies sample rates that we associate with music
// and thus eligible for better resampling and fast capture.
// This is somewhat less than 44100 to allow for pitch correction
// involving resampling as well as asynchronous resampling.
#define AUDIO_PROCESSING_MUSIC_RATE 40000

static inline bool isMusicRate(uint32_t sampleRate) {
  return sampleRate >= AUDIO_PROCESSING_MUSIC_RATE;
}

} // namespace cocos2d

// ---------------------------------------------------------------------------
